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Network Working Group                                       J. RosenbergRequest for Comments: 4168                                 Cisco SystemsCategory: Standards Track                                 H. Schulzrinne                                                     Columbia University                                                            G. Camarillo                                                                Ericsson                                                            October 2005The Stream Control Transmission Protocol (SCTP)as a Transport for the Session Initiation Protocol (SIP)Status of This Memo   This document specifies an Internet standards track protocol for the   Internet community, and requests discussion and suggestions for   improvements.  Please refer to the current edition of the "Internet   Official Protocol Standards" (STD 1) for the standardization state   and status of this protocol.  Distribution of this memo is unlimited.Copyright Notice   Copyright (C) The Internet Society (2005).Abstract   This document specifies a mechanism for usage of SCTP (the Stream   Control Transmission Protocol) as the transport mechanism between SIP   (Session Initiation Protocol) entities.  SCTP is a new protocol that   provides several features that may prove beneficial for transport   between SIP entities that exchange a large amount of messages,   including gateways and proxies.  As SIP is transport-independent,   support of SCTP is a relatively straightforward process, nearly   identical to support for TCP.Rosenberg, et al.           Standards Track                     [Page 1]

RFC 4168              SCTP as a Transport for SIP           October 2005Table of Contents1. Introduction ....................................................22. Terminology .....................................................23. Potential Benefits ..............................................23.1. Advantages over UDP ........................................33.2. Advantages over TCP ........................................34. Transport Parameter .............................................55. SCTP Usage ......................................................55.1. Mapping of SIP Transactions into SCTP Streams ..............56. Locating a SIP Server ...........................................67. Security Considerations .........................................78. IANA Considerations .............................................79. References ......................................................79.1. Normative References .......................................79.2. Informative References .....................................81.  Introduction   The Stream Control Transmission Protocol (SCTP) [4] has been designed   as a new transport protocol for the Internet (or intranets) at the   same layer as TCP and UDP.  SCTP has been designed with the transport   of legacy SS7 signaling messages in mind.  We have observed that many   of the features designed to support transport of such signaling are   also useful for the transport of SIP (the Session Initiation   Protocol) [5], which is used to initiate and manage interactive   sessions on the Internet.   SIP itself is transport-independent, and can run over any reliable or   unreliable message or stream transport.  However, procedures are only   defined for transport over UDP and TCP.  This document defines   transport of SIP over SCTP.2.  Terminology   The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",   "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this   document are to be interpreted as described inRFC 2119 [1].3.  Potential BenefitsRFC 3257 presents some of the key benefits of SCTP [10].  We   summarize some of these benefits here and analyze how they relate to   SIP (a more detailed analysis can be found in [12]).Rosenberg, et al.           Standards Track                     [Page 2]

RFC 4168              SCTP as a Transport for SIP           October 20053.1.  Advantages over UDP   All the advantages that SCTP has over UDP regarding SIP transport are   also shared by TCP.  Below, there is a list of the general advantages   that a connection-oriented transport protocol such as TCP or SCTP has   over a connection-less transport protocol such as UDP.   Fast Retransmit: SCTP can quickly determine the loss of a packet,      because of its usage of SACK and a mechanism that sends SACK      messages faster than normal when losses are detected.  The result      is that losses of SIP messages can be detected much faster than      when SIP is run over UDP (detection will take at least 500 ms, if      not more).  Note that TCP SACK exists as well, and TCP also has a      fast retransmit option.  Over an existing connection, this results      in faster call setup times under conditions of packet loss, which      is very desirable.  This is probably the most significant      advantage of SCTP for SIP transport.   Congestion Control: SCTP maintains congestion control over the entire      association.  For SIP, this means that the aggregate rate of      messages between two entities can be controlled.  When SIP is run      over TCP, the same advantages are afforded.  However, when run      over UDP, SIP provides less effective congestion control.  This is      because congestion state (measured in terms of the UDP retransmit      interval) is computed on a transaction-by-transaction basis,      rather than across all transactions.  Thus, congestion control      performance is similar to opening N parallel TCP connections, as      opposed to sending N messages over one TCP connection.   Transport-Layer Fragmentation: SCTP and TCP provide transport-layer      fragmentation.  If a SIP message is larger than the MTU size, it      is fragmented at the transport layer.  When UDP is used,      fragmentation occurs at the IP layer.  IP fragmentation increases      the likelihood of having packet losses and makes NAT and firewall      traversal difficult, if not impossible.  This feature will become      important if the size of SIP messages grows dramatically.3.2.  Advantages over TCP   We have shown the advantages of SCTP and TCP over UDP.  We now   analyze the advantages of SCTP over TCP.   Head of the Line: SCTP is message-based, as opposed to TCP, which is      stream-based.  This allows SCTP to separate different signalling      messages at the transport layer.  TCP only understands bytes.      Assembling received bytes to form signalling messages is performed      at the application layer.  Therefore, TCP always delivers anRosenberg, et al.           Standards Track                     [Page 3]

RFC 4168              SCTP as a Transport for SIP           October 2005      ordered stream of bytes to the application.  On the other hand,      SCTP can deliver signalling messages to the application as soon as      they arrive (when using the unordered service).  The loss of a      signalling message does not affect the delivery of the rest of the      messages.  This avoids the head of line blocking problem in TCP,      which occurs when multiple higher layer connections are      multiplexed within a single TCP connection.  A SIP transaction can      be considered an application layer connection.  There are multiple      transactions running between proxies.  The loss of a message in      one transaction should not adversely effect the ability of a      different transaction to send a message.  Thus, if SIP is run      between entities with many transactions occurring in parallel,      SCTP can provide improved performance over SIP over TCP (but not      SIP over UDP; SIP over UDP is not ideal from a congestion control      standpoint; see above).   Easier Parsing: Another advantage of message-based protocols, such as      SCTP and UDP, over stream-based protocols, such as TCP, is that      they allow easier parsing of messages at the application layer.      There is no need to establish boundaries (typically using      Content-Length headers) between different messages.  However, this      advantage is almost negligible.   Multihoming: An SCTP connection can be associated with multiple IP      addresses on the same host.  Data is always sent over one of the      addresses, but if it becomes unreachable, data sent to one can      migrate to a different address.  This improves fault tolerance;      network failures making one interface of the server unavailable do      not prevent the service from continuing to operate.  SIP servers      are likely to have substantial fault tolerance requirements.  It      is worth noting that, because SIP is message oriented and not      stream oriented, the existing SRV (Service Selection) procedures      defined in [5] can accomplish the same goal, even when SIP is run      over TCP.  In fact, SRV records allow the 'connection' to fail      over to a separate host.  Since SIP proxies can run statelessly,      failover can be accomplished without data synchronization between      the primary and its backups.  Thus, the multihoming capabilities      of SCTP provide marginal benefits.   It is important to note that most of the benefits of SCTP for SIP   occur under loss conditions.  Therefore, under a zero loss condition,   SCTP transport of SIP should perform on par with TCP transport.   Research is needed to evaluate under what loss conditions the   improvements in setup times and throughput will be observed.Rosenberg, et al.           Standards Track                     [Page 4]

RFC 4168              SCTP as a Transport for SIP           October 20054.  Transport Parameter   Via header fields carry a transport protocol identifier.RFC 3261   defines the value "SCTP" for SCTP, but does not define the value for   the transport parameter for TLS over SCTP.  Note that the value   "TLS", defined byRFC 3261, is intended for TLS over TCP.   Here we define the value "TLS-SCTP" for the transport part of the Via   header field to be used for requests sent over TLS over SCTP [8].   The updated augmented BNF (Backus-Naur Form) [2] for this parameter   is the following (the original BNF for this parameter can be found inRFC 3261):   transport         =  "UDP" / "TCP" / "TLS" / "SCTP" / "TLS-SCTP"                        / other-transport   The following are examples of Via header fields using "SCTP" and   "TLS-SCTP":     Via: SIP/2.0/SCTP ws1234.example.com:5060     Via: SIP/2.0/TLS-SCTP ws1234.example.com:50605.  SCTP Usage   Rules for sending a request over SCTP are identical to TCP.  The only   difference is that an SCTP sender has to choose a particular stream   within an association in order to send the request (seeSection 5.1).   Note that no SCTP identifier needs to be defined for SIP messages.   Therefore, the Payload Protocol Identifier in SCTP DATA chunks   transporting SIP messages MUST be set to zero.   The SIP transport layers of both peers are responsible for managing   the persistent SCTP connection between them.  On the sender side, the   core or a client (or server) transaction generates a request (or   response) and passes it to the transport layer.  The transport sends   the request to the peer's transaction layer.  The peer's transaction   layer is responsible for delivering the incoming request (or   response) to the proper existing server (or client) transaction.  If   no server (or client) transaction exists for the incoming message,   the transport layer passes the request (or response) to the core,   which may decide to construct a new server (or client) transaction.5.1.  Mapping of SIP Transactions into SCTP Streams   SIP transactions need to be mapped into SCTP streams in a way that   avoids Head Of the Line (HOL) blocking.  Among the different ways of   performing this mapping that fulfill this requirement, we have chosenRosenberg, et al.           Standards Track                     [Page 5]

RFC 4168              SCTP as a Transport for SIP           October 2005   the simplest one; a SIP entity SHOULD send every SIP message (request   or response) over stream zero with the unordered flag set.  On the   receiving side, a SIP entity MUST be ready to receive SIP messages   over any stream.      In the past, it was proposed that SCTP stream IDs be used as      lightweight SIP transaction identifiers.  That proposal was      withdrawn because SIP now provides (as defined inRFC 3261 [5]) a      transaction identifier in the branch parameter of the Via entries.      This transaction identifier, missing in the previous SIP spec [9],      makes it unnecessary to use the SCTP stream IDs to demultiplex SIP      traffic.   In many circumstances, SIP requires the use of TLS [3], for instance,   when routing a SIPS URI [5].  As defined inRFC 3436 [8], TLS running   over SCTP MUST NOT use the SCTP unordered delivery service.   Moreover, any SIP use of an extra layer between the transport layer   and SIP that requires ordered delivery of messages MUST NOT use the   SCTP unordered delivery service.   SIP applications that require ordered delivery of messages from the   transport layer (e.g., TLS) SHOULD send SIP messages belonging to the   same SIP transaction over the same SCTP stream.  Additionally, they   SHOULD send messages belonging to different SIP transactions over   different SCTP streams, as long as there are enough available   streams.      A common scenario where the above mechanism should be used      consists of two proxies exchanging SIP traffic over a TLS      connection using SCTP as the transport protocol.  This works      because all of the SIP transactions between the two proxies can be      established within one SCTP association.   Note that if both sides of the association follow this   recommendation, when a request arrives over a particular stream, the   server is free to return responses over a different stream.  This   way, both sides manage the available streams in the sending   direction, independently of the streams chosen by the other side to   send a particular SIP message.  This avoids undesirable collisions   when seizing a particular stream.6.  Locating a SIP Server   The primary issue when sending a request is determining whether the   next hop server supports SCTP so that an association can be opened.   SIP entities follow normal SIP procedures to discover [6] a server   that supports SCTP.Rosenberg, et al.           Standards Track                     [Page 6]

RFC 4168              SCTP as a Transport for SIP           October 2005   However, in order to use TLS on top of SCTP, an extra definition is   needed.RFC 3263 defines the NAPTR (Naming Authority Pointer) [7]   service value "SIP+D2S" for SCTP, but fails to define a value for TLS   over SCTP.  Here we define the NAPTR service value "SIPS+D2S" for   servers that support TLS over SCTP [8].7.  Security Considerations   The security issues raised inRFC 3261 [5] are not worsened by SCTP,   provided the advice inSection 5.1 is followed and TLS over SCTP [8]   is used where TLS would be required inRFC 3261 [5] or inRFC 3263   [6].  So, the mechanisms described inRFC 3436 [8] MUST be used when   SIP runs on top of TLS [3] and SCTP.8.  IANA Considerations   This document defines a new NAPTR service field value (SIPS+ D2S).   The IANA has registered this value under the "Registry for the SIP   SRV Resource Record Services Field".  The resulting entry is as   follows:   Services Field        Protocol  Reference   --------------------  --------  ---------   SIPS+D2S              SCTP      [RFC4168]9.  References9.1.  Normative References   [1]  Bradner, S., "Key words for use in RFCs to Indicate Requirement        Levels",BCP 14,RFC 2119, March 1997.   [2]  Crocker, D. and P. Overell, "Augmented BNF for Syntax        Specifications: ABNF",RFC 2234, November 1997.   [3]  Dierks, T. and C. Allen, "The TLS Protocol Version 1.0",RFC2246, January 1999.   [4]  Stewart, R., Xie, Q., Morneault, K., Sharp, C., Schwarzbauer,        H., Taylor, T., Rytina, I., Kalla, M., Zhang, L., and V. Paxson,        "Stream Control Transmission Protocol",RFC 2960, October 2000.   [5]  Rosenberg, J., Schulzrinne, H., Camarillo, G., Johnston, A.,        Peterson, J., Sparks, R., Handley, M., and E. Schooler, "SIP:        Session Initiation Protocol",RFC 3261, June 2002.   [6]  Rosenberg, J. and H. Schulzrinne, "Session Initiation Protocol        (SIP): Locating SIP Servers",RFC 3263, June 2002.Rosenberg, et al.           Standards Track                     [Page 7]

RFC 4168              SCTP as a Transport for SIP           October 2005   [7]  Mealling, M., "Dynamic Delegation Discovery System (DDDS) Part        Three: The Domain Name System (DNS) Database",RFC 3403, October        2002.   [8]  Jungmaier, A., Rescorla, E., and M. Tuexen, "Transport Layer        Security over Stream Control Transmission Protocol",RFC 3436,        December 2002.9.2.  Informative References   [9]  Handley, M., Schulzrinne, H., Schooler, E., and J. Rosenberg,        "SIP: Session Initiation Protocol",RFC 2543, March 1999.   [10] Coene, L., "Stream Control Transmission Protocol Applicability        Statement",RFC 3257, April 2002.   [11] Camarillo, G., "The Internet Assigned Number Authority (IANA)        Uniform Resource Identifier (URI) Parameter Registry for the        Session Initiation Protocol (SIP)",BCP 99,RFC 3969, December        2004.   [12] Camarillo, G., Schulrinne, H., and R. Kantola, "Evaluation of        Transport Protocols for the Session Initiation Protocol", IEEE,        Network vol. 17, no. 5, 2003.Rosenberg, et al.           Standards Track                     [Page 8]

RFC 4168              SCTP as a Transport for SIP           October 2005Authors' Addresses   Jonathan Rosenberg   Cisco Systems   600 Lanidex Plaza   Parsippany, NJ  07054   US   Phone: +1 973 952-5000   EMail: jdrosen@cisco.com   URI:http://www.jdrosen.net   Henning Schulzrinne   Columbia University   M/S 0401   1214 Amsterdam Ave.   New York, NY  10027-7003   US   EMail: schulzrinne@cs.columbia.edu   Gonzalo Camarillo   Ericsson   Hirsalantie 11   Jorvas  02420   Finland   EMail: Gonzalo.Camarillo@ericsson.comRosenberg, et al.           Standards Track                     [Page 9]

RFC 4168              SCTP as a Transport for SIP           October 2005Full Copyright Statement   Copyright (C) The Internet Society (2005).   This document is subject to the rights, licenses and restrictions   contained inBCP 78, and except as set forth therein, the authors   retain all their rights.   This document and the information contained herein are provided on an   "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS   OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET   ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,   INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE   INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED   WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.Intellectual Property   The IETF takes no position regarding the validity or scope of any   Intellectual Property Rights or other rights that might be claimed to   pertain to the implementation or use of the technology described in   this document or the extent to which any license under such rights   might or might not be available; nor does it represent that it has   made any independent effort to identify any such rights.  Information   on the procedures with respect to rights in RFC documents can be   found inBCP 78 andBCP 79.   Copies of IPR disclosures made to the IETF Secretariat and any   assurances of licenses to be made available, or the result of an   attempt made to obtain a general license or permission for the use of   such proprietary rights by implementers or users of this   specification can be obtained from the IETF on-line IPR repository athttp://www.ietf.org/ipr.   The IETF invites any interested party to bring to its attention any   copyrights, patents or patent applications, or other proprietary   rights that may cover technology that may be required to implement   this standard.  Please address the information to the IETF at ietf-   ipr@ietf.org.Acknowledgement   Funding for the RFC Editor function is currently provided by the   Internet Society.Rosenberg, et al.           Standards Track                    [Page 10]